[asterisk-users] manager event privilege: call, all? what is?
what mean privilege: call, all for event? this is call or all? why have two privilege And how i can prevent manager to send me NewExt event?___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ztdummy and Asterisk
Hi to all, The problem was finally solved by installing Asterisk-1.4.18. Versions 1.4.20 and current (1.4.22) produce the issue. Version 1.4.18 does not. May this be a bug? Aldo Sudak___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is SIPPEER curcalls working for you ?
Title: Re: [asterisk-users] Is SIPPEER curcalls working for you ? Did You triedhttp://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers? Hi, In this threadhttp://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html, I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1 (Whenever Bob is calling Alice, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 1 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 0) Would you say it's expected ? I would say SIPPEER:curcalls should include both inbound and outbound calls. I also added a fourth Daniel SIP Phone. Whenever Alice is calling Bob and Daniel is calling Alice, SIPPEER(Alice:curcalls) equals 1 (I would wait 2) SIPPEER(Bob:curcalls) equals 1 (I would wait 1) SIPPEER(Daniel:curcalls) equals 0 (I would wait 0) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-962 Asterisk
Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn¹t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that is a bitch when you have many phones, shouldn¹t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?
http://bugs.digium.com/view.php?id=13645
Hi list,
I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...
Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation and forwards them to B, also over IAX.
When B hangs up, it transmits IAX hangup which A receives
who, in turn, does not transmit the IAX hangup to its user
X, Y or Z. So X, Y or Z still think the call is up...
All of this is verified with iax debug... A receives the hangup but
never hangs up the other side if running 1.4.22. Everything is ok
if running 1.4.21.2.
Could this be something we're doing wrong ? What steps would
you suggest for further diagnostic?
Thanks in advance for any feedback.
System A runs 1.4.22 / 1.4.21.2
System A iax.conf
[userX]
type=user
transfer=no
host=dynamic
secret=whatever
context=the-context
disallow=all
allow=alaw
allow=ulaw
[systemB]
type=peer
qualify=200
transfer=no
host=ip-here
disallow=all
allow=gsm
System A extensions.conf:
[the-context]
exten = _.,1,Wait(1)
exten = _.,n,Set(CALL_UUID=${EXTEN})
exten = _.,n,Set(RESULT_STRING=${ODBC_CALL_DATA_4_UUID(${CALL_UUID})})
exten = _.,n,Set(ARRAY(NAME,ACCT,IAXUSER,NUM)=${RESULT_STRING})
exten = _.,n,Set(DONT_CARE=${ODBC_REMOVE_CALL_4_UUID(${CALL_UUID})})
exten = _.,n,Set(CALLERID(name)=${NAME})
exten = _.,n,Set(CDR(accountcode)=${ACCT})
exten = _.,n,Dial(IAX2/[EMAIL PROTECTED]/${NUM})
exten = _.,n,Hangup()
(note: behaviour is also failing in 1.4.22 if, instead of Dialing
system B, we just wait+hangup directly here!)
System B runs asterisk 1.2.30.1
System B iax.conf:
[one-systemA-user]
type=user
context=one-context
notransfer=yes
disallow=all
allow=gsm
System B extensions.conf:
[one-context]
exten = _N,1,Dial(.../${EXTEN})
exten = _N,n,Hangup()
--
exvito
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Re: [asterisk-users] SPA-962 Asterisk
I've never used the Sipura phones but they probably sync with an NTP server. My guess is that the NTP server is on the asterisk box (you can probably verify this by checking the config of the phones and finding the option for NTP server). It is possible that the NTP service isn't running on the asterisk box (after a reboot or a crash) or that the asterisk box's time is incorrect. Do you know what distribution you are running on the server? You can type 'uname -a' at a command prompt and get an idea of the distro. Also try '/etc/init.d/ntpd' start or 'service ntpd start' - these may be able to restart the NTP daemon for you and begin syncing the phones properly again. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Anness Sent: Tuesday, November 04, 2008 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA-962 Asterisk Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn't right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that is a bitch when you have many phones, shouldn't the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards, device compatibility, etc. The conference call will be held as usual on the Talkshoe service for people calling in from normal (G.711) phones. The Talkshoe bridge can be reached by PSTN or SIP URI. Anyone with G.722 capable phones (some models of Polycom, Snom, Cisco, Avaya, Mitel, Siemens or Grandstream) or a G.722 capable soft phone (Eyebeam, OEM version only) will be able to connect to the ZipDX conference bridge and participate in glorious wideband audio. The two conference bridges will be connected. People connected to ZipDX directly will be able to hear the startling difference that HDVoice makes. This is especially true in conference calls where line quality, accents and background noise all cause intelligibility issues. The downloadable recording of the conference will let everyone hear the difference for themselves. The call will happen Friday Nov 7 at 12 noon EST. To find out more about how to join this call please visit: http://blog.mgraves.org/ or http://voipusersconference.org/ning/ Michael Graves -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is SIPPEER curcalls working for you ? [SOLVED]
2008/11/4 Igor Zamocky [EMAIL PROTECTED] Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers ? I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1 (Whenever Bob is calling Alice, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 1 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 0) Would you say it's expected ? I would say SIPPEER:curcalls should include both inbound and outbound calls. I also added a fourth Daniel SIP Phone. Whenever Alice is calling Bob and Daniel is calling Alice, SIPPEER(Alice:curcalls) equals 1 (I would wait 2) SIPPEER(Bob:curcalls) equals 1 (I would wait 1) SIPPEER(Daniel:curcalls) equals 0 (I would wait 0) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, Nov 04, 2008 at 09:34:18AM -0600, Michael Graves wrote: This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards, device compatibility, etc. The conference call will be held as usual on the Talkshoe service for people calling in from normal (G.711) phones. The Talkshoe bridge can be reached by PSTN or SIP URI. Anyone with G.722 capable phones (some models of Polycom, Snom, Cisco, Avaya, Mitel, Siemens or Grandstream) or a G.722 capable soft phone (Eyebeam, OEM version only) will be able to connect to the ZipDX conference bridge and participate in glorious wideband audio. Is there any decent free soft phone that is not capable of Speex/wb? A short check on my system: supporting: ekiga 2.0.12-1+nmu1 linphone 2.1.1-1+b1 twinkle1:1.2-3 Not supporting: iaxcomm2.0.2-3 But then again, this is just IaxComm. And even it has some tweaks to speex. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, 4 Nov 2008 17:48:58 +0200, Tzafrir Cohen wrote: Is there any decent free soft phone that is not capable of Speex/wb? A short check on my system: supporting: ekiga 2.0.12-1+nmu1 linphone 2.1.1-1+b1 twinkle1:1.2-3 Not supporting: iaxcomm2.0.2-3 But then again, this is just IaxComm. And even it has some tweaks to speex. The trouble is that there is extremely limited hardware support for Speex. G.722 is freely available and has better hardware support. Did anyone notice yesterdays press release from Audio Codes announcing broad wideband support across their product range by some time in 2009? In any case, the wideband bridge for this weeks VUC call supports only G.722. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 Time on Asterisk
On 4 Nov 2008, at 14:17, Steve Anness wrote: Duplicated stuff Just the one post is usual sufficient. On topic - I would check what the NTP server is set to on the SPA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call terminates after 20 minutes
Hi, It does not appears to be a session-timers issue. There are no SIP exchange except for BYE message initited after 20 minutes. I increased the session timer to 3600 seconds and also tried your suggestion without any luck. Any other inputs? Thanks Jim On Mon, Nov 3, 2008 at 10:47 PM, John Todd [EMAIL PROTECTED] wrote: Go to sip.conf. Find the SIP Session-Timers section. Ensure that you have this option set: session-timers=refuse This might help. If not, try other variations of the session-timers value. The default session-timer is 10 minutes - exactly half of what you claim is your duration maximums, so it seems suspiciously like that might have something to do with it. Maybe not. In any case, fire up wireshark/tethereal and watch the SIP packets for a particular call to see what's happening - distrust everything other than what you see on the wire and then work backwards. An understanding of SIP packet flows will be helpful here, or the ladder view of SIP transactions that is built into wireshark's graphical interface will certainly help as well. JT On Nov 2, 2008, at 11:07 AM, Jim Boykin wrote: Any help. Thanks On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin [EMAIL PROTECTED] wrote: Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: Jim Boykin pisze: We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Sesion-timer set but not supported by sip-peers? --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote: In any case, the wideband bridge for this weeks VUC call supports only G.722. But we do plan to make a recording of both conference version available, AFAIK? r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
randulo wrote: On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote: In any case, the wideband bridge for this weeks VUC call supports only G.722. But we do plan to make a recording of both conference version available, AFAIK? r But will it be a high-def recording? N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, 4 Nov 2008 17:10:42 +0100, randulo wrote: On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote: In any case, the wideband bridge for this weeks VUC call supports only G.722. But we do plan to make a recording of both conference version available, AFAIK? r Absolutely. We should have a wideband recording that plainly illustrates the difference between callers on G.711 vs G.722. Also, a narrowband recording. It'll be interesting to hear if the people connected via G.722 sound appreciably better in the narrowband recording. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] users.conf and hints
Is there a way to override sip peers defined in users.conf with respect to their context and hints? Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. Thanks for any help. nurscarepbx*CLI core show version Asterisk 1.4.22 built by root @ nurscarepbx on a x86_64 running Linux on 2008-10-16 12:37:36 UTC Pbx*CLI show hints ... [EMAIL PROTECTED] : SIP/1203 State:Idle Watchers 0 [EMAIL PROTECTED] : SIP/1202 State:Idle Watchers 0 ... Users.conf [1202] fullname = 1202 secret = 1202 hasvoicemail = yes mailbox = [EMAIL PROTECTED] vmsecret = 1234 hassip = yes hasmanager = no callwaiting = no context = from-nortel call-limit = 4 dynamic = yes qualify = yes host = dynamic [1203] fullname = 1203 secret = 1203 hasvoicemail = yes mailbox = [EMAIL PROTECTED] vmsecret = 1234 hassip = yes hasmanager = no callwaiting = no context = from-nortel subscribecontext = internal call-limit = 4 dynamic = yes qualify = yes host = dynamic Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, 4 Nov 2008, Michael Graves wrote: Also, a narrowband recording. It'll be interesting to hear if the people connected via G.722 sound appreciably better in the narrowband recording. Will you be able to calibrate volume and frequency response (within G.711 limits) between the conferences to reduce subjective influences? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
Dear List I'm asking if there is a small hardware already implemented with software that just do traffic Shaping QoS? I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer *But it does not look available in Italy ! Regards Andrea ** Kristian Kielhofner ha scritto: On Jan 9, 2008 11:44 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Erik Anderson wrote: On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote: Heh yeah that's what I was thinking of doing. What's the traffic shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Sweet. We had been using HTB but upgraded all our CPE to use HFSC when AstLinux did and found it great. Matt, I've been looking for feedback on HFSC... I don't get to play around as much as I used to and I STILL haven't been able to really compare HTB and HFSC for VoIP traffic. What have you found? How do you like the other changes to AstShape (iptables CLASSIFY, etc)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, 4 Nov 2008 08:42:29 -0800 (PST), Steve Edwards wrote: On Tue, 4 Nov 2008, Michael Graves wrote: Also, a narrowband recording. It'll be interesting to hear if the people connected via G.722 sound appreciably better in the narrowband recording. Will you be able to calibrate volume and frequency response (within G.711 limits) between the conferences to reduce subjective influences? My understanding is that the wideband capable bridge includes such provisions. It can bridge wideband and narrowband channels making all the necessary adjustments. One of the things that we hope to discuss is the internals of conference bridges. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 Time on Asterisk
Steve Anness wrote: Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Asterisk does not push time to the phones. The phones themselves will contact an NTP server to obtain the correct time. Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn’t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that isn’t the best solution when you have many phones, shouldn’t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Best practice is to configure ntpd as both a server and client on your Asterisk server and point your phones to it. That way your phones will have exactly the same time as your Asterisk server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, Nov 4, 2008 at 10:34 AM, Michael Graves [EMAIL PROTECTED] wrote: This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards, device compatibility, etc. The conference call will be held as usual on the Talkshoe service for people calling in from normal (G.711) phones. The Talkshoe bridge can be reached by PSTN or SIP URI. Anyone with G.722 capable phones (some models of Polycom, Snom, Cisco, Avaya, Mitel, Siemens or Grandstream) or a G.722 capable soft phone (Eyebeam, OEM version only) will be able to connect to the ZipDX conference bridge and participate in glorious wideband audio. The two conference bridges will be connected. People connected to ZipDX directly will be able to hear the startling difference that HDVoice makes. This is especially true in conference calls where line quality, accents and background noise all cause intelligibility issues. The downloadable recording of the conference will let everyone hear the difference for themselves. The call will happen Friday Nov 7 at 12 noon EST. To find out more about how to join this call please visit: http://blog.mgraves.org/ or http://voipusersconference.org/ning/ Michael Graves While I appreciate the benefits of wideband audio/G722/etc and your efforts to educate people about it during the VoIP User's Conference, I wince at the thought of hearing a sales pitch. At the very least I find some of this company's existing marketing literature suspect (big surprise there, right): ZipDX is the only audio conferencing service available today that can support the HD Voice capabilities of the Polycom IP 6000 and IP 7000. Our system was designed from the ground up to take full advantage of these devices. You can also experience HDVoice on Polycom's SoundPoint(R) IP-550, IP-560 and IP-650. taken from: http://www.zipdx.com/Info/PolycomSS another gem: Patented No-Codes Conferencing Apparently they've patented dialing out to conference participants and/or reading Caller ID/ANI to bypass the pin. More fantastic work from the US Patent Office! I know there is more to HD Voice than G.722 but I also know that at least (in the FOSS world alone) FreeSWITCH supports conferencing in G.722. Pingtel's sipx uses FreeSWITCH as a conference engine. I would bet with some investigation various other commercial products support it too. This was originally posted to various Asterisk-related groups and I certainly realize FreeSWITCH isn't Asterisk. I just doubt that the CEO from a company that makes questionable claims such as this is going to contribute much to what is (usually) an otherwise interesting discussion about (largely) FOSS technology. More than likely you're going to get a sales pitch for their hosted service, which at $0.10/min per participant is expensive. I understand they offer other conferencing features but it's pretty clear they are leading with their exclusive support for wideband audio. I'm sorry for the sarcastic tone and poor attitude but I'm getting increasingly frustrated with sales and marketing goons infiltrating what are otherwise excellent opportunities for *purely* technical discussions. Asterisk-Biz is overrun, they appear on Asterisk-Users from time to time, and they've all but taken over most conferences. Come on guys, don't let it happen to the VoIP User's Conference!! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
Kristian, heh... In a word: They've got a great product. Whether you actually want to go read ANY company's BS ^H^H web site is another story! I've rarelty if ever seen a web site with anything other than gobbledy gook text, but so few people read the text in a web site, it matters little. As far as the codecs, I think David Franken knows a lot about this technology and I look forward to hearing more form him. ZipDX have got what they're trying to do right, the semantics are a whole other question; Best, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some progress, anyway...
Just saw from build 2036: Starting mini_httpd... WARNING WARNING WARNING YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN INTERFACE IMMEDIATELY! Example: htpasswd /var/www/admin/.htpasswd admin WARNING WARNING WARNING Starting mini_httpd (HTTP only)... cat: can't open '/tmp/mydhcpip': No such file or directory This is pbx (Linux i586 2.6.25.19-astlinux) 11:30:27 pbx login: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some progress, anyway...
Philip Prindeville wrote: Just saw from build 2036: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe Whoops. I'm sure Philip thought he was sending this to a different mailing list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is SIPPEER curcalls working for you ?
Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1 (Whenever Bob is calling Alice, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 1 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 0) Would you say it's expected ? I would say SIPPEER:curcalls should include both inbound and outbound calls. I also added a fourth Daniel SIP Phone. Whenever Alice is calling Bob and Daniel is calling Alice, SIPPEER(Alice:curcalls) equals 1 (I would wait 2) SIPPEER(Bob:curcalls) equals 1 (I would wait 1) SIPPEER(Daniel:curcalls) equals 0 (I would wait 0) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some progress, anyway...
Darrick Hartman wrote: Philip Prindeville wrote: Just saw from build 2036: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe Whoops. I'm sure Philip thought he was sending this to a different mailing list. Indeed... another infamous (well, not quite) autocompletion blooper. Never mind. -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the best way to resale termination/origination?
Hi, We have been selling * systems for a while and always have used other companies for origination and termination and let the client pay directly. Since we do not have enough traffic to justify building our own infrastructure we would like to start reselling someone else's service. Any ideas? It would have to include easy way for billing. Thank you. robert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi trunk does not compile with kernel 2.6.27
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and it will not compile with this kernel whereas it does compile with 2.6.25. Here is the relevant portion of the build: VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_3_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_4_30 HOSTCC /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes.o HOSTLD /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes GEN /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes CHECK /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 Use of uninitialized value in concatenation (.) or string at /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 line 74. make[3]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes.verified] Error 1 make[2]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp] Error 2 make[1]: *** [_module_/usr/src/dahdi-trunk/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.27-gentoo-r2' make: *** [modules] Error 2 Any assistance on this would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blank Voicemail.Conf after Password Change
On Tuesday 04 November 2008 12:37:03 Leah Newmark wrote: Thanks! Apparently, I forgot to mention our version running is 1.2.13 Is the coding similar enough to give it a shot? No, the code is likely to be completely different. In fact, the trouble you're having may just be the reason why it was changed. (Perhaps it's time to upgrade...) I would advise you to strongly consider it. Not only are legacy bugs not getting fixed anymore in that series, but that version is exceedingly vulnerable to multiple security issues. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi trunk does not compile with kernel 2.6.27
OK, thanks.
on Tuesday 11/04/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is the relevant portion of the build:
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_3_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_4_30
HOSTCC /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes.o
HOSTLD /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes
GEN /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes
CHECK /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
Use of uninitialized value in concatenation (.) or string at
/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 line 74.
make[3]: ***
[/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes.verified]
Error 1
make[2]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp] Error 2
make[1]: *** [_module_/usr/src/dahdi-trunk/drivers/dahdi] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.27-gentoo-r2'
make: *** [modules] Error 2
Any assistance on this would be appreciated.
A known issue: http://bugs.digium.com/13832
'Use of uninitialized value $ENV{XBUS_NAME}' if stderr from make
redirected to a file
I hope to get it fixed. In the mean time: patch out that test from
drivers/dahdi/xpp/Kbuild
--
Tzafrir Cohen
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+972-50-7952406 mailto:[EMAIL PROTECTED]
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How do
you spend it?
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[EMAIL PROTECTED]
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Re: [asterisk-users] Call terminates after 20 minutes
Other things to check:
- ensure you don't have media timeouts set (see rtptimeout in
sip.conf)
- do you have any absolute timers set in your dialplan? (any $
{TIMEOUT(absolute)} values set?
- does your upstream carrier have any sort of timer limit? (try
using just a vanilla softphone to call directly to your trunking
provider, no Asterisk involved.)
What does your console say in full debug mode?
JT
On Nov 4, 2008, at 8:04 AM, Jim Boykin wrote:
Hi,
It does not appears to be a session-timers issue. There are no SIP
exchange except for BYE message initited after 20 minutes.
I increased the session timer to 3600 seconds and also tried your
suggestion without any luck. Any other inputs?
Thanks
Jim
On Mon, Nov 3, 2008 at 10:47 PM, John Todd [EMAIL PROTECTED] wrote:
Go to sip.conf.
Find the SIP Session-Timers section.
Ensure that you have this option set:
session-timers=refuse
This might help. If not, try other variations of the session-timers
value. The default session-timer is 10 minutes - exactly half of
what
you claim is your duration maximums, so it seems suspiciously like
that might have something to do with it. Maybe not. In any case,
fire up wireshark/tethereal and watch the SIP packets for a
particular
call to see what's happening - distrust everything other than what
you
see on the wire and then work backwards. An understanding of SIP
packet flows will be helpful here, or the ladder view of SIP
transactions that is built into wireshark's graphical interface will
certainly help as well.
JT
On Nov 2, 2008, at 11:07 AM, Jim Boykin wrote:
Any help. Thanks
On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin [EMAIL PROTECTED]
wrote:
Marcin, can you elaborate. No timer has been set and call is not
idle either.
Thanks
Jim
On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk
[EMAIL PROTECTED] wrote:
Jim Boykin pisze:
We are running Asterisk SVN. We are facing a strange and
repetable
problem. All outgoing call gets terminated in approx 20 minutes.
Asterisk initiates BYE message to the remote end and call
terminates.
Sesion-timer set but not supported by sip-peers?
---
John Todd
[EMAIL PROTECTED]+1-256-428-6083
Asterisk Open Source Community Director
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Re: [asterisk-users] SPA-962 Asterisk
On Nov 4, 2008, at 6:08 AM, Steve Anness wrote: Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn’t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that is a bitch when you have many phones, shouldn’t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group Your SPA devices almost certainly get their timing data from an NTP server. Some devices will find their NTP servers with DHCP options requests, and I think there is even a Zeroconf method for determining local NTP servers, but I doubt anyone uses that method. One of the more novel methods I used a while ago (when I was doing design for ATAs) was to use the Date: header in the SIP INVITE as the time set. The device would know it's GMT offset, and then calculate what time it was. It was an ugly hack, and that ATA was the reason that the Date: header is in INVITEs in Asterisk. :-) I was kind of ashamed of it a while ago (NTP is much more correct solution) but the more pragmatic I get in my old age, the more elegant it seems. No NTP stack to worry about, no additional firewall holes, etc. etc. and phones really don't need to be super-accurate so NTP is typically overkill anyway. ATAs especially, since they don't even display the date - they just send it along with the Caller ID data when a new call event is generated. JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi trunk does not compile with kernel 2.6.27
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is the relevant portion of the build:
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_3_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_4_30
HOSTCC /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes.o
HOSTLD /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes
GEN /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes
CHECK /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
Use of uninitialized value in concatenation (.) or string at
/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 line 74.
make[3]: ***
[/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes.verified]
Error 1
make[2]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp] Error 2
make[1]: *** [_module_/usr/src/dahdi-trunk/drivers/dahdi] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.27-gentoo-r2'
make: *** [modules] Error 2
Any assistance on this would be appreciated.
A known issue: http://bugs.digium.com/13832
'Use of uninitialized value $ENV{XBUS_NAME}' if stderr from make
redirected to a file
I hope to get it fixed. In the mean time: patch out that test from
drivers/dahdi/xpp/Kbuild
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] OT: Traffic Shaping
On 5/11/2008 5:57 a.m., andrea wrote: Dear List I'm asking if there is a small hardware already implemented with software that just do traffic Shaping QoS? I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer *But it does not look available in Italy ! Regards Andrea The Linksys WRTG routers do, and anything that you can run DDWRT or OpenWRT on. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shared voicemail box
Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. [EMAIL PROTECTED] (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 interdigit timers bug?
Am Montag, den 03.11.2008, 13:20 + schrieb Steve Davies: I found this only last week... The problem is not the short timer, it is the dialtone audio definition (top of the same page IIRC). If you look at the tone definition for Dialtone it is only a few seconds long. When it runs out, the call is disconnected. I have lengthened our dialtone pattern to 50 seconds, and the long timer to 60 seconds, which works here. Steve, thanks for your answer, but it didn't work for me. This is my tone definition: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2) which means, as far as I understand, to play both(1+2) 350Hz at -19Db and 440Hz at -19Db, 10 cycles, permanently(*), with no silence laps(0). I tried changing the cycles number 10 up to 500, just to see what happens, but has no effect on timers definition. Changing the short timer has effect, but works also as the long timer... Any more ideas? -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?
On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=13645 Thanks Igor, we'll keep an eye on it. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shared voicemail box
Simply put the shared mailbox on all the phones definition like: mailbox=100,101 That way the phone will flash if there is mail on any of the boxes. On Tue, 2008-11-04 at 13:31 -0800, Kelvin Chan wrote: Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. [EMAIL PROTECTED] (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
Did you know that any commandyou type in asterisk cli starting with exclamation point (!) is execute in the shell by asterisk ?? Example : running !ls will run 'ls' in your current directory So, be aware because your user can do whatever we want then. Dima wrote: On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote: Setting the user's shell to /usr/sbin/rasterisk works. On login user gets into asterisk CLI if asterisk is running (user just has to have write permission to /var/lib/asterisk.*). How does that user login? client$ ssh [EMAIL PROTECTED] password: Asterisk SVN-branch-1.4-r137138, Copyright (C) 1999 - 2008 Digium, Inc. and others. ... Verbosity is at least 9 asterisk.machine*CLI CLI has the ability to create extensions, extensions which could execute the System application, pick up his phone, dial the extension, execute the command, and even cover his tracks by putting NoCDR in the extension path and removing the incriminating extension afterwards (again with the CLI). In 1.4, it's even easier: he can originate a call from the command line, perhaps even to a phone of a person he wanted to take the fall for his exploit. The person I'm giving the access to is an admin of that asterisk. It's up to him to do evil stuff with asterisk itself. as long as he can't get a shell and do rm -rf / I'm safe. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blank Voicemail.Conf after Password Change
Thanks! Apparently, I forgot to mention our version running is 1.2.13 Is the coding similar enough to give it a shot? (Perhaps it's time to upgrade...) I really appreciate the assistance. LN [EMAIL PROTECTED] wrote: Message: 12 Date: Mon, 3 Nov 2008 15:11:10 -0600 From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] Blank Voicemail.Conf after Password Change To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Monday 03 November 2008 12:19:31 Leah Newmark wrote: The directory is owned by asterisk and permissions seem fine there too: drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 . I never see a voicemail.conf.new created. I have done a locate on the whole server and don't see a blank one there either which might be causing any confusion. In 1.4, it no longer creates any such file. Instead, it creates a randomly named file, designed to be unique, starting with voicemail.conf. and ending with 6 random characters. ps xaguwww confirms asterisk is running as UID asterisk: asterisk 24560 1.0 3.1 50648 32692 ?Ssl Oct02 506:35 /usr/sbin/asterisk -U asterisk I see that it's been up for a while, and I'm wondering if that coincides with when we started noticing this behavior. I'd have to restart asterisk to use strace, but restarting for all I know might help... Any other input before I do so? Please try the following patch: http://bugs.digium.com/view.php?id=13831 -- Tilghman -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Tue, 4 Nov 2008, Dima wrote: The person I'm giving the access to is an admin of that asterisk. It's up to him to do evil stuff with asterisk itself. as long as he can't get a shell and do rm -rf / I'm safe. Hmm, I wonder if you could run asterisk in a jail? Anyone done that on FreeBSD for example? That would solve your issues I think. It would certainly be difficult for your admin to admin asterisk without the CLI. Depending on your flavor of GUI it may be difficult for him to admin asterisk with shell access. Without a jail, however, if you give him CLI access you are basically giving him the machine, which seems to be the general consensus. Has anyone ever tried to compile ! out of the CLI? j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-962 Time on Asterisk
Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn¹t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that isn¹t the best solution when you have many phones, shouldn¹t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote: Setting the user's shell to /usr/sbin/rasterisk works. On login user gets into asterisk CLI if asterisk is running (user just has to have write permission to /var/lib/asterisk.*). How does that user login? client$ ssh [EMAIL PROTECTED] password: Asterisk SVN-branch-1.4-r137138, Copyright (C) 1999 - 2008 Digium, Inc. and others. ... Verbosity is at least 9 asterisk.machine*CLI CLI has the ability to create extensions, extensions which could execute the System application, pick up his phone, dial the extension, execute the command, and even cover his tracks by putting NoCDR in the extension path and removing the incriminating extension afterwards (again with the CLI). In 1.4, it's even easier: he can originate a call from the command line, perhaps even to a phone of a person he wanted to take the fall for his exploit. The person I'm giving the access to is an admin of that asterisk. It's up to him to do evil stuff with asterisk itself. as long as he can't get a shell and do rm -rf / I'm safe. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels are increasing without limit - Please Help!
It was a lack of free space in disk, because a big load of recorded calls and logs. Daniel On Thu, Oct 23, 2008 at 12:40 PM, Daniel - Asterisk [EMAIL PROTECTED]wrote: I'm restarting my system without solution and I've extended my call limit to 10 calls (asterisk.conf) to avoid call rectriction. But, why now? It was working well from July until this morning. Thanks in advance for every help you can give. Daniel On Thu, Oct 23, 2008 at 12:04 PM, Daniel - Asterisk [EMAIL PROTECTED]wrote: My version is 1.4.21.1 On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED] wrote: Suddenly my system crash whem I see core show channels are increasing until reaches its limit at asterisk.conf It seems channels (Local, Zap, SIP) are not being closed. The problem persists and I don't know what to do Please help me! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Tuesday 04 November 2008 15:52:10 Ruddy Gbaguidi wrote: Did you know that any commandyou type in asterisk cli starting with exclamation point (!) is execute in the shell by asterisk ?? Example : running !ls will run 'ls' in your current directory So, be aware because your user can do whatever we want then. Yes, but remote commands are executed as whatever user is running the remote command, which is NOT necessarily the same as root. You can open up the permissions of the asterisk.ctl pipe file to allow another group to connect. That, however, still leaves the indirect method of executing commands, which are still executed by the Asterisk process itself. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Tuesday 04 November 2008 16:02:40 Jeff LaCoursiere wrote: On Tue, 4 Nov 2008, Dima wrote: The person I'm giving the access to is an admin of that asterisk. It's up to him to do evil stuff with asterisk itself. as long as he can't get a shell and do rm -rf / I'm safe. Hmm, I wonder if you could run asterisk in a jail? Anyone done that on FreeBSD for example? That would solve your issues I think. It would certainly be difficult for your admin to admin asterisk without the CLI. Depending on your flavor of GUI it may be difficult for him to admin asterisk with shell access. Without a jail, however, if you give him CLI access you are basically giving him the machine, which seems to be the general consensus. Even with a jail, you are giving a user complete control of the capabilities of the user that Asterisk is running as. Period. There is no way around this. If Asterisk is running as root, then giving CLI access is the same as giving complete control of your machine over to anybody with CLI access. Has anyone ever tried to compile ! out of the CLI? As I stated before, this does not improve your security one iota. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] users.conf and hints
On Tuesday 04 November 2008 10:30:10 Jeremy Mann wrote: Is there a way to override sip peers defined in users.conf with respect to their context and hints? No, there is not. Users.conf is meant to be a very simple interface for adding users, designed especially for the Asterisk GUI project. It has never been a general purpose good interface for the full complement of Asterisk features. If you want a simple interface, use users.conf. If you want full features, use each configuration file separately. Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. That's not what subscribecontext does. Subscribecontext contains the context where your phone will LOOK for extension states when it watches other extensions. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING message when calls get into a queue with realtime members (Local channel)
Hi, I'm using queue configuration as follows: - queues from* queues.conf* - queue_members from *external Database thru ODBC*, using* Local channels * as interface - sip extensions from *external Database thru ODBC* When a call is sent from queue to an interface (local channel), it is answered but a message appears at the CLI: *[Nov 4 16:56:04] WARNING[13951]: app_queue.c:3014 try_calling: The device state of this queue member, Ag-Daniel, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. * The interface value for agent *Ag-Daniel* is: *Local/[EMAIL PROTECTED]/n*and final interface is *SIP/7070* I have checked the UPGRADE.txt file and set *limitonpeer=yes *at sip.conf and every sip_buddie has *call-limit=2 *and *type=friend *but problem persists. My asterisk version is: 1.4.21.1 Calls are entering well now, but what when load increases high? Does anyone know what can I do to avoid this WARNINGS and future issues? Thanks again, Daniel Arohuanca +51 1 994149553 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sendmail using SMTP authorization
Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getting it to work. I've included the following below: Asterisk 1.4.21 CentOS 5 Sendmail 8.13.8 === bounced mail === === maillog === === hosts === === access === === authinfo === === sendmail.mc === The bounced mail file shows the authentication problem, although there's also a troubling DSN: Service unavailable message that appears in maillog. I'm not sure whether the two are related or if the latter is really a problem at all. Any help would be welcome. Thanks in advance! Cheers, Hugh CentOS 5 Sendmail 8.13.8 === bounced mail === = From [EMAIL PROTECTED] Sun Nov 2 11:53:57 2008 Return-Path: [EMAIL PROTECTED] Received: from localhost (localhost) by rapperyo.com (8.13.8/8.13.8) id mA2Gru4B002917; Sun, 2 Nov 2008 11:53:56 -0500 Date: Sun, 2 Nov 2008 11:53:56 -0500 From: Mail Delivery Subsystem [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: multipart/report; report-type=delivery-status; boundary=mA2Gru4B002917.1225644836/rapperyo.com Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --mA2Gru4B002917.1225644836/rapperyo.com The original message was received at Sun, 2 Nov 2008 11:53:56 -0500 from rapperyo.com [127.0.0.1] - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html) - Transcript of session follows - ... while talking to smtp-rog.mail.yahoo.com.: MAIL From:[EMAIL PROTECTED] 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html 554 5.0.0 Service unavailable --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/delivery-status Reporting-MTA: dns; rapperyo.com Received-From-MTA: DNS; rapperyo.com Arrival-Date: Sun, 2 Nov 2008 11:53:56 -0500 Final-Recipient: RFC822; [EMAIL PROTECTED] Action: failed Status: 5.0.0 Diagnostic-Code: SMTP; 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html Last-Attempt-Date: Sun, 2 Nov 2008 11:53:56 -0500 --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/rfc822 Return-Path: [EMAIL PROTECTED] Received: from rapperyo.com (rapperyo.com [127.0.0.1]) by rapperyo.com (8.13.8/8.13.8) with ESMTP id mA2Gru4B002915 for [EMAIL PROTECTED]; Sun, 2 Nov 2008 11:53:56 -0500 Received: (from [EMAIL PROTECTED]) by rapperyo.com (8.13.8/8.13.8/Submit) id mA2GrtoD002914; Sun, 2 Nov 2008 11:53:55 -0500 Date: Sun, 2 Nov 2008 11:53:55 -0500 From: root [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: I'm sending mail from the Terminal! --mA2Gru4B002917.1225644836/rapperyo.com-- === maillog === Nov 2 11:49:35 pbx sendmail[2421]: alias database /etc/aliases rebuilt by root Nov 2 11:49:35 pbx sendmail[2421]: /etc/aliases: 76 aliases, longest 10 bytes, 765 bytes total Nov 2 11:49:35 pbx sendmail[2426]: starting daemon (8.13.8): SMTP [EMAIL PROTECTED]:00:00 Nov 2 11:49:35 pbx sm-msp-queue[2434]: starting daemon (8.13.8): [EMAIL PROTECTED]:00:00 Nov 2 11:53:56 pbx sendmail[2914]: mA2GrtoD002914: from=root, size=71, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], [EMAIL PROTECTED] Nov 2 11:53:56 pbx sendmail[2915]: mA2Gru4B002915: from=[EMAIL PROTECTED], size=318, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=rapperyo.com [127.0.0.1] Nov 2 11:53:56 pbx sendmail[2914]: mA2GrtoD002914: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=30071, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (mA2Gru4B002915 Message accepted for delivery) Nov 2 11:53:56 pbx sendmail[2917]: mA2Gru4B002915: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=120318, relay=smtp- rog.mail.yahoo.com. [206.190.36.18], dsn=5.0.0, stat=Service unavailable Nov 2 11:53:56 pbx sendmail[2917]: mA2Gru4B002915: mA2Gru4B002917: DSN: Service unavailable Nov 2 11:53:57 pbx sendmail[2917]: mA2Gru4B002917: to=[EMAIL PROTECTED], delay=00:00:01, xdelay=00:00:01, mailer=local, pri=31546, dsn=2.0.0, stat=Sent === hosts === === # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 rapperyo.com pbx.local pbx localhost.localdomain localhost 192.168.2.160 www.rapperyo.com ::1 localhost6.localdomain6 localhost6 === access === # Check the /usr/share/doc/sendmail/README.cf file for a description # of the format of this
Re: [asterisk-users] Sendmail using SMTP authorization
Try using SSMTP http://www.linux.com/articles/132006 It works with any provider for mail sending, and takes 30 seconds to setup. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hugolivude Sent: Tuesday, November 04, 2008 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Sendmail using SMTP authorization Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getting it to work. I've included the following below: Asterisk 1.4.21 CentOS 5 Sendmail 8.13.8 === bounced mail === === maillog === === hosts === === access === === authinfo === === sendmail.mc === The bounced mail file shows the authentication problem, although there's also a troubling DSN: Service unavailable message that appears in maillog. I'm not sure whether the two are related or if the latter is really a problem at all. Any help would be welcome. Thanks in advance! Cheers, Hugh CentOS 5 Sendmail 8.13.8 === bounced mail === = From [EMAIL PROTECTED] Sun Nov 2 11:53:57 2008 Return-Path: [EMAIL PROTECTED] Received: from localhost (localhost) by rapperyo.com (8.13.8/8.13.8) id mA2Gru4B002917; Sun, 2 Nov 2008 11:53:56 -0500 Date: Sun, 2 Nov 2008 11:53:56 -0500 From: Mail Delivery Subsystem [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: multipart/report; report-type=delivery-status; boundary=mA2Gru4B002917.1225644836/rapperyo.com Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --mA2Gru4B002917.1225644836/rapperyo.com The original message was received at Sun, 2 Nov 2008 11:53:56 -0500 from rapperyo.com [127.0.0.1] - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html) - Transcript of session follows - ... while talking to smtp-rog.mail.yahoo.com.: MAIL From:[EMAIL PROTECTED] 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html 554 5.0.0 Service unavailable --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/delivery-status Reporting-MTA: dns; rapperyo.com Received-From-MTA: DNS; rapperyo.com Arrival-Date: Sun, 2 Nov 2008 11:53:56 -0500 Final-Recipient: RFC822; [EMAIL PROTECTED] Action: failed Status: 5.0.0 Diagnostic-Code: SMTP; 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html Last-Attempt-Date: Sun, 2 Nov 2008 11:53:56 -0500 --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/rfc822 Return-Path: [EMAIL PROTECTED] Received: from rapperyo.com (rapperyo.com [127.0.0.1]) by rapperyo.com (8.13.8/8.13.8) with ESMTP id mA2Gru4B002915 for [EMAIL PROTECTED]; Sun, 2 Nov 2008 11:53:56 -0500 Received: (from [EMAIL PROTECTED]) by rapperyo.com (8.13.8/8.13.8/Submit) id mA2GrtoD002914; Sun, 2 Nov 2008 11:53:55 -0500 Date: Sun, 2 Nov 2008 11:53:55 -0500 From: root [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: I'm sending mail from the Terminal! --mA2Gru4B002917.1225644836/rapperyo.com-- === maillog === Nov 2 11:49:35 pbx sendmail[2421]: alias database /etc/aliases rebuilt by root Nov 2 11:49:35 pbx sendmail[2421]: /etc/aliases: 76 aliases, longest 10 bytes, 765 bytes total Nov 2 11:49:35 pbx sendmail[2426]: starting daemon (8.13.8): SMTP [EMAIL PROTECTED]:00:00 Nov 2 11:49:35 pbx sm-msp-queue[2434]: starting daemon (8.13.8): [EMAIL PROTECTED]:00:00 Nov 2 11:53:56 pbx sendmail[2914]: mA2GrtoD002914: from=root, size=71, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], [EMAIL PROTECTED] Nov 2 11:53:56 pbx sendmail[2915]: mA2Gru4B002915: from=[EMAIL PROTECTED], size=318, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=rapperyo.com [127.0.0.1] Nov 2 11:53:56 pbx sendmail[2914]: mA2GrtoD002914: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=30071, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (mA2Gru4B002915 Message accepted for delivery) Nov 2 11:53:56 pbx sendmail[2917]: mA2Gru4B002915: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=120318, relay=smtp- rog.mail.yahoo.com. [206.190.36.18], dsn=5.0.0, stat=Service unavailable
Re: [asterisk-users] MS Exchange IMAP Voicemail
On Sun, Oct 5, 2008 at 8:04 PM, David Backeberg [EMAIL PROTECTED] wrote: Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would it be different? When I setup my voicemail.conf for IMAP Asterisk does not work right. sip show peers only shows 1 peer. The CLI is freezing up, etc. When I turn off the IMAP voicemail these problems go away. I configured everything how it should be so I am wondering if someone can post a configuration they know works 100% with Exchange IMAP server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Tue, Nov 04, 2008 at 04:02:40PM -0600, Jeff LaCoursiere wrote: Hmm, I wonder if you could run asterisk in a jail? Anyone done that on FreeBSD for example? That would solve your issues I think. It would certainly be difficult for your admin to admin asterisk without the CLI. Depending on your flavor of GUI it may be difficult for him to admin asterisk with shell access. I don't think Asterisk is a good candidate for chrooting. It re-reads the config files in /etc on each reload. It will occasionally rotates logs in /var/log/asterisk . Just to mention a few. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get
On Tue, Nov 04, 2008 at 04:31:58PM -0600, Tilghman Lesher wrote: On Tuesday 04 November 2008 15:52:10 Ruddy Gbaguidi wrote: Did you know that any commandyou type in asterisk cli starting with exclamation point (!) is execute in the shell by asterisk ?? Example : running !ls will run 'ls' in your current directory So, be aware because your user can do whatever we want then. Yes, but remote commands are executed as whatever user is running the remote command, which is NOT necessarily the same as root. You can open up the permissions of the asterisk.ctl pipe file to allow another group to connect. '!' is not a remote command. If you login as asteriskcli and asterisk is running as the user asteriskd, '!ls' and '!rm whatever' will be executed through /bin/sh by the user asteriskcli . Anything you can cause Asterisk to run through the dialplan, originate and such would be run by asteriskd. So it doesn't buy you much vs. creating a standard user account. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] twice normal beep before busy tone ??
Hi Rob, Also try without the r option to the dial command: http://www.voip-info.org/wiki-Asterisk+cmd+dial Rob After I removed the r option I now nearly immediately get the busy tone. BUT: The wiki says Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however. What does appropriate mean?? After removing the r Asterisk starts playing the ring tone, when the called phone starts ringing. When the called phone is a cell phone, this takes up to 3 seconds. During this time, there is silence! Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
